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[Incoming Call] Mediation Server responds PRACK with 481 RRS feed

  • Question

  • Hi everybody,

    the connection from the Mediation Server to the PSTN is realized through a SIP Trunk. In this case, the mediation server is directly connected with the SBC by the provider. in a Test scenario a PSTN subscriber calls  the ocs user (MOC). After the SBC has sent a PRACK request to the mediation server, the server responds with 481 and then followed with a 408 Request Time out. The PSTN caller hears a reject tone after a while. In my opinion, the host part (ProviderDomain) in To header field could be a problem for the Mediation Server because he knows only the IP address of the SBC as the next hop but he does not know the domain of the provider. Perhaps it would make sense that  the host part  in To header field the IP address or the domain of Mediation server (MyDomain) contains. Or is my suspicion wrong and the problem lies elsewhere? May be the cause is as the RFC 3262/Section 3 it describes, and what will that mean in this case? See SIP Trace.

    PS: Outgoing Calls are working properly

    OCS 2007 R2

    OCS User: +331234567

    PSTN User: +331234568

    SBC -> OCS
    ----------------------------------------
    Jul  9 15:44:04.761 SBC:Port sent to Mediation:5060
    INVITE sip:+331234567@Mediation SIP/2.0
    Via: SIP/2.0/TCP SBC:5060;branch=z9hG4bKekvubh104oa1ln671.1
    To: <sip:+331234567@ProviderDomain;user=phone>
    From: "01234568" <sip:+331234568@ProviderDomain;user=phone>;tag=SDtrcjc01-87d114f
    Call-ID: SDtrcjc01-b3219ca979490a62819ba464f164-iikc131
    CSeq: 1 INVITE
    Max-Forwards: 68
    Contact: <sip:01234568@SBC:5060;transport=tcp>
    Date: Fri, 9 Jul 2010 15:44:04 GMT
    Allow: INVITE, ACK, PRACK, CANCEL, BYE, OPTIONS, MESSAGE, NOTIFY, UPDATE, REGISTER, INFO, REFER, SUBSCRIBE
    Supported: 100rel
    P-Asserted-Identity: <sip:01234568@ProviderDomain>
    Accept: application/sdp, application/isup, application/xml, application/dtmf-relay
    Content-Type: application/sdp
    Content-Length: 236
    v=0
    o=- 0 59079 IN IP4 SBC
    s=IMSS
    c=IN SBC
    t=0 0
    m=audio 56215 RTP/AVP 8 101 18 106
    a=rtpmap:101 G726-32/8000
    a=rtpmap:106 telephone-event/8000
    a=sendrecv
    a=ptime:20
    a=sqn:0
    a=cdsc: 1 image udptl t38

    OCS answers with 100 Trying
    ----------------------------------------

    OCS -> SBC
    ----------------------------------------
    Jul  9 15:44:04.782 On SBC:Port received from Mediation:5060
    SIP/2.0 183 Session Progress
    FROM: "01234568"<sip:+331234568@ProviderDomain;user=phone>;tag=SDtrcjc01-87d114f
    TO: <sip:+331234567@ProviderDomain;user=phone>;tag=7e3c73eab;epid=B21D0D1D2
    CSEQ: 1 INVITE
    CALL-ID: SDtrcjc01-b3219ca979490a62819ba464f164-iikc131
    VIA: SIP/2.0/TCP SBC:5060;branch=z9hG4bKekvubh104oa1ln671.1
    CONTACT: <sip:med.MyDomain:5060;transport=Tcp;maddr=IP-Mediation>
    CONTENT-LENGTH: 260
    CONTENT-TYPE: application/sdp; charset=utf-8
    ALLOW: UPDATE
    REQUIRE: 100rel
    SERVER: RTCC/3.5.0.0 MediationServer
    Rseq: 1
    v=0
    o=- 80 1 IN IP4 Mediation
    s=session
    c=IN IP4 Mediation
    b=CT:1000
    t=0 0
    m=audio 63941 RTP/AVP 8 106
    c=IN IP4 Mediation
    a=rtcp:63941
    a=label:Audio
    a=rtpmap:8 PCMA/8000
    a=rtpmap:106 telephone-event/8000
    a=fmtp:106 0-16
    a=ptime:20

    SBC -> OCS
    ----------------------------------------
    Jul  9 15:44:04.788 On SBC:Port sent to Mediation:5060
    PRACK sip:med.MyDomain:5060;maddr=IP-Mediation;transport=Tcp SIP/2.0
    Via: SIP/2.0/TCP SBC:5060;branch=z9hG4bKf460eq10507ovp2m1.1
    To: <sip:+331234567@ProviderDomain;user=phone>;tag=7e3c73eab;epid=B21D0D1D2
    From: "01234568"<sip:+331234568@ProviderDomain;user=phone>;tag=SDtrcjc01-87d114f
    Call-ID: SDtrcjc01-b3219ca979490a62819ba464f164-iikc131
    CSeq: 2 PRACK
    Max-Forwards: 68
    RAck: 1  1 INVITE
    Content-Length: 0

    Ocs -> SBC
    ----------------------------------------
    Jul  9 15:44:04.790 On SBC:Port received from Mediation:5060
    SIP/2.0 481 Call Leg/Transaction Does Not Exist
    From: "01234568"<sip:+331234568@ProviderDomain;user=phone>;tag=SDtrcjc01-87d114f
    To: <sip:+331234567@ProviderDomain;user=phone>;tag=7e3c73eab;epid=B21D0D1D2
    CSEQ: 2 PRACK
    CALL-ID: SDtrcjc01-b3219ca979490a62819ba464f164-iikc131
    VIA: SIP/2.0/TCP SBC:5060;branch=z9hG4bKf460eq10507ovp2m1.1
    CONTENT-LENGTH: 0
    SERVER: RTCC/3.5.0.0 MediationServer

    Ocs -> SBC
    ----------------------------------------
    Jul  9 15:44:38.233 On SBC:Port received from Mediation:5060
    SIP/2.0 408 Request Timeout
    From: "01234568"<sip:+331234568@ProviderDomain;user=phone>;tag=SDtrcjc01-87d114f
    To: <sip:+331234567@ProviderDomain;user=phone>;tag=7e3c73eab;epid=B21D0D1D2
    CSEQ: 1 INVITE
    CALL-ID: SDtrcjc01-b3219ca979490a62819ba464f164-iikc131
    VIA: SIP/2.0/TCP SBC:5060;branch=z9hG4bKekvubh104oa1ln671.1
    CONTENT-LENGTH: 0
    SERVER: RTCC/3.5.0.0 MediationServer

    Thanks in advance!

    Friday, July 16, 2010 1:55 PM

Answers

  • Hi Randy.

    First thanks for your answer.

    I have spoken with ISP and they said that they will make a SIP manipulation on the SBC. They have replaced the host part in the To header field with the IP address (Gateway Listening IP address) of the mediation server and the Incoming calls from PSTN work now.

    Thanks again for your help!

    • Marked as answer by Reyane Monday, July 26, 2010 1:48 PM
    Monday, July 26, 2010 1:48 PM

All replies

  • Is your SIP trunk provider listed on the OIP page? The reason I ask, is that it certifies compatability.

     

    If not, you can still probably get it to work with some help from your provider.

     

    Can you confirm from your provider the Audio codec they are using for your trunk? I think I saw G726 in one of the traces. Should be G711

    If you have the flexibility to, I would suggest looking into a provider on the OIP page, it will eliminate all of the headaches here with a non certified provider.

     

    http://technet.microsoft.com/en-us/office/bb735838.aspx

    Thanks.


    Randy Wintle | MCTS: UC Voice Specialization | Winxnet Inc
    Friday, July 23, 2010 7:27 PM
  • Hi Randy.

    First thanks for your answer.

    I have spoken with ISP and they said that they will make a SIP manipulation on the SBC. They have replaced the host part in the To header field with the IP address (Gateway Listening IP address) of the mediation server and the Incoming calls from PSTN work now.

    Thanks again for your help!

    • Marked as answer by Reyane Monday, July 26, 2010 1:48 PM
    Monday, July 26, 2010 1:48 PM